1. Field of the Invention
This invention relates generally to transmission and switching techniques in telephone communication systems and, more particularly, to an improved conference technique whereby a number of channels in a telephone switching system employing pulse code modulation for transmission purposes are combined so that a number of subscribers may participate in a common telephone conversation. More particularly still, it relates to improvements in a three-port conference circuit of the type disclosed in U.S. Pat. Nos. 3,699,264 and 4,007,338, both of which are assigned to the same assignee as the present invention.
The present invention pertains to a three-port conference circuit for use in a private automatic branch exchange similar to those units manufactured by GTE Automatic Electric Incorporated and designated GTD120. Circuitry with minimum modification could also be employed in class five central offices that employ digital switching. Such telephone systems employ a time switching network rather than the most prevalent earlier space divided switching network.
In time division switching networks a requirement exists to have sources of pulse code modulated voice samples associated with time slots. These time slots allow the conference to sequentially receive the code for each conferee. For the conference circuit to be effective, it must be able to recognize who the conferees are and, of course, who is not associated with the conference. The circuitry must also be capable of distributing the conference speaker's code to each conferee. Information of this sort is, of course, available in the telephone switching systems referred to above. It should be understood that only telephone switching systems employing pulse code modulation can use the circuitry of the present invention, and such circuitry interfaces with time division portions of such switching networks. Other codes (linear) could be used, but a modified decision algorithum would have to be used, since in the present disclosure, D2/D3 type coding is required.
2. Description of the Prior Art
An approach to the handling of pulse code modulated information and conference circuitry is taught by U.S. Pat. Nos. 3,699,264 and 4,007,338, which are assigned to the same assignee as the present invention. In these noted patents, digital signals are not converted to analog; rather binary words are compared from the participating channels, with the smallest binary numbers (this corresponds to the largest analog signal) selected as the speaker. Various improvements in the conference circuitry disclosed in these above-identified U.S. patents are disclosed in U.S. Pat. Nos. 4,002,981 and 4,054,755, both of which are assigned to the same assignee as the present invention.
PCM conferencing as taught in the above-identified patents and application requires a source of pulse code modulated (PCM) coded voice samples which have associated time slots. These time slots allow the conference to sequentially receive a code for each conferee. The conference circuitry must be able to recognize who the conferees are and who is not associated with the conference call. The conference circuit, in U.S. Pat. No. 4,007,338, then determines the loudest PCM voice sample during each PCM time frame, storing and outputing the selected PCM code to all conferees. In other words, the binary words are compared from the participating channels, with the smallest binary members selected as the speaker. These patents, therefore, utilize a minimum binary code to select the speaker. This technique, called "instant speaker selection", for generating conferencing, however, is subject to degradation due to the presence of idle channel noise and from non-talking conferees with individual circuit DC offset voltage variations, as the speaker's audio signal passes through the region and its PCM sample is at a high weight value. In addition, when two or more conference members are conversing simultaneously, the conference circuit could alternately select a new speaker for each time frame, thus degrading the quality of the speech of conversing conferees.
In U.S. Pat. No. 4,022,981, an improved multi-port (beyond 3) conference circuit is disclosed utilizing a minimum binary code as employed in the coding formats (D2 and D3) currently employed in pulse code modulated telephony. Generally, the method for choosing the speaker is to clear the PCM buffers at time slot 94. Then the first conferee detected is loaded to a conferee register. The register is compared to a temporary speaker register. If the conferee code corresponds to a larger pulse amplitude modulated (PAM) sample (that is, it presents a smaller binary value PCM code), the conferee code is loaded into the temporary speaker register. Each new conferee code is loaded to the conferee register and then compared to the temporary speaker register. If it is smaller in binary value but larger in PAM, it is then transferred and becomes the new temporary speaker. If not, it is written over when the next conferee code is loaded in. Finally, time slot 94 occurs and the temporary speaker register is transferred to a conference speaker register. This then is the conference speaker for the next frame, and this register contains the PCM code which all conferees except the speaker himself will receive. It will then be updated one frame later during the next occurrence of time slot 94.
In U.S. Pat. No. 4,054,755, a further improvement in the multi-port conference circuitry is provided. These improvements attempt to solve the idle channel noise and circuit offset variation problem and, also, foreign signal protection (i.e., 60Hz signal longitudinally coupled to the line). In the conference circuit, PCM samples are taken for each conferee from the time switch and, via comparator circuits, a PCM sample is sent to the conferee. Since the selected PCM sample is not determined until all samples are compared, a frame delay is required after which all conferees except the selected conferee will receive the selected PCM sample from the previous frame. The selected conferee, in turn, receives a null code (perfect idle channel). To minimize speech clipping or selecting noise, two circuits, a preliminary and a preferred speaker preference circuit, are employed.
The preliminary preference circuit utilizes the identity of the previous selected speaker and after its PCM sample is compared, its binary weight is modified to the highest value of a corresponding curve segment. This is done by adding a bit between the segment and the step bits, allowing the binary value to be decreased. This technique permits the conference circuit to hold on to the previous speaker if the incoming PCM samples are in the same PCM segment or below in value.
The preferred speaker preference circuit functions when the magnitude of the present PCM sample exceeds the value of the preferred preference circuit threshold. When a speaker is selected for the succeeding frame and has a larger PAM (smaller PCM code) sample than the threshold, a preferred preference circuit creates a lower binary weight (apparently larger PAM) to the comparator, for the selected speaker, for a period of one frame. This reduces speech clipping during that time when two or more conferees are conversing simultaneously.
Neither the preliminary nor the preferred preference circuits alters the incoming or the outgoing PCM sample to the conferees; additional binary weights are only presented to the comparison circuit to favor the previous speaker.